Re: [Asterisk-Users] One Way Audio....in the middle of a call
I experienced this today. Doing a 'show channels' in Asterisk showed a Zap line perpetually ringing the sip phone even though the sip phone was reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip allowed 2-way audio to resume. Phil Frederic Jean wrote: Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; you will see if they change port or something like that after a while. Cheers, Frederic - Original Message - *From:* Geoff Manning mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Tuesday, April 25, 2006 17:37 *Subject:* [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Vonage
Something I've been curious about is if it is possible to stick their ata on a extra ethernet port on an Asterisk server and have the Asterisk server spoof the Vonage server. Then, do a man-in-the-middle type thing to use the ata for authentication, but have Asterisk handle all the calls. Perhaps another idea is to hammer an ata with authentication requests and create a long list of nonces and hashes that you replay back to the server as needed. Phil mustardman29 wrote: You MUST have a softphone account. Some blog said they may open it up to all accounts at some point in the future but that is only a rumor right now. -Original Message- From: Steve Jones [mailto:[EMAIL PROTECTED] Sent: Thursday, March 30, 2006 4:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk with Vonage OK - I should have been more specific in my original post, but before I get too excited to try this tonight when I get home, does this work with ANY vonage line (ie: first, unlimited line) or does it have to be on a softphone enabled line? Thanks!! From: Adrian A [mailto:[EMAIL PROTECTED] Sent: Wed 3/29/2006 9:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Vonage Works great with these settings: [vonage] type=peer secret=password username=phone number host=sphone.vopr.vonage.net port=5061 dtmfmode=rfc2833 fromuser=phone number fromdomain=sphone.vopr.vonage.net canreinvite=no context=vonage_incoming insecure=very On 3/29/06, Steve Jones [EMAIL PROTECTED] wrote: I know Vonage doesn't officially have a bring your own device type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Toshiba Strata DK-280 support?
Charles Marcus wrote: [...] So, how much work are we talking about to get our current system to play nice with Asterisk? Will we lose any functionality? Gain any? Do you know of any technical how-to's that my phone guy would be able to answer these questions from? Are you available to concult? If so, for how much? Sorry to hit you with so much, but if I don't ask... ;) We have a DK40 system. The simpliest integration was to yank our ancient voicemail box and replace it with Asterisk (which is set up to act seemlessly as the new voicemail server). In addition, it can also route calls, patch VOIP calls through, allow for some test voip phone extensions, etc. It was a pretty simple way for us to get our feet wet. It cost a server (which we had) and a 4-port FXO card (I think it was $300). Off the bat, it became a new voicemail system with new features our old system didn't have (like emailling voicemails as attachments). Over time it's done much more. At some point (in a few months, probably) we'll turn off the Toshiba and put viop phones on everyone's desk (including some people's at a remote office and homes). It should also cut our phone bill down to a 1/10th of what it is now! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
Omar A. Sabek wrote: Like BJ, I'm sorry you had bad luck Phil. I have been playing with this phone all weekend, and I have had minor problems. The voice quality is as good as my cisco and polycom sip phones. I asked a friend to guess what kind of phone I was talking on and he said it sounded like a regular home or office phone. I have been very happy with the voice quality. My first day was a huge disappointment. Three crashes, calls wouldn't work over my work's wifi (eventhough it registered ok), short battery time, lost settings after a crash, etc. However, after I went in and cleared my settings back to default, the troubles went away! I'm been using it for over three days without a glitch. So, I would recommend to anybody else who is getting one of these phones, to immediately set all settings back to 'default' (under the Tools menu) before spending too much time configuring it. I reported on the voip-info page dismal talk times but it must have been an anomoly. Today I spoke for over an hour on the phone and still had plenty of juice left. My battery life seems to have improved as well. I don't know if that's was a glitch fixed by setting things back to the defaults, or if cycling the battery is helping. I also have less of a tendency to play with the menus, and the backlight could be a power drainer (it is quite bright). All-in-all this phone is a winner. It works with Asterisk flawlessly. As long as my troubles don't come back, I would agree. I think my phone was shipped to me in a funny state causing it not to work right. It's a winner now. There are some little things I would wish for, but I'm quite happy with it. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys WIP300 WiFi Phone
Philip Edelbrock wrote: Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. After it charged and I started using it, I had three crashes. Once during a call (exactly 3 minutes into it, according to the frozen display), and twice while the phone wasn't in use. When I woke the phone up it had a blank white display. To unhang the phone, I removed the battery. My settings were lost after two of the crashes (it's possible that the first crash was before I had any significant settings and so didn't notice them gone). With so many settings (multiple wifi configs, sip configs, email, phone preferences, etc.) it is quite painful to have to start over. Are there others who have this phone? Have you had it crash like this? I'm wondering if this is a firmware issue, or if I have a defective phone. :'( Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linksys WIP300 WiFi Phone
Whoo hoo! I just received my WIP300 from voipsupply. I have to let it charge before I can play with it. A few quick comments: - I started a Wiki page at voip-info to post issues, firmware news, etc. I really like the wealth of info on the GXP-2000 page, so I wanted to start something similar for this phone. http://www.voip-info.org/wiki/index.php?page=Linksys%20WIP300 - My kit didn't come with a CD-ROM or registration card, eventhough they are listed as being in the Package Contents. - This phone uses a USB port to charge, do firmware updates, and perhaps other things. Sadly... it DOES NOT COME WITH A USB CORD! You'd think for $250 that you'd get a cord included... oh well. It does come with a charger with a usb end so you can charge the phone from an AC outlet, though. - The battery charging animation runs backwards, animating like the battery's charge is flowing out rather than in. A little amusing. No charge status while it is charging, which I don't like. It would be nice to see that it's, say, 75% charged for example. Does anyone else have one of these phones yet? Any gotcha's as far as using it with *? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sniffing sip password/uri/host info
Rich Adamson wrote: I want to sniff all these info to test a sip ip phone talking to a asterisk server. I have used tcpdump, but It just shows the Ethereal would probably be a batter analyzer. Not sure how well it seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you can also get tcpdump to dump raw data and analyze it off-line with ethereal. Ethereal does a pretty good job at decoding both sip and iax packets. I use it a lot (on a separate laptop). Try Cane and Abel. It automaticly grabs sip registration info and categories it for you. Records the calls to sound files, too. http://www.oxid.it/cain.html Very easy to use. Can even try to crack the password hashes if you want. (If you can wait a few years ;') You can also turn on sip debugging from the CLI which will dump out the headers for you. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400P's in a single machine
Marc Archer wrote: Can someone give me a definite answer as to wether or not you can reliably run multiple TDM400P’s in the same machine? I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key system, but I have seen several threads suggesting that this is not a supported configuration Potentially easier and cheaper is to just have the FXO's and connect it as analog extensions. That's what we did here. It serves as a replacement voicemail system, and IVR while still being able to answer and route calls, provide gateway to voip extensions, etc. Less waste, too, if/when you completely switch to voip. FXO ports are likely to be more usable than FXS if that becomes the case, I would imagine. That said, I think some folks have run multiple TDM400's by making sure they have unique IRQ's. I'll let somebody else speak to the details of that. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application Faxing using SIP
On Feb 18, 2006, at 11:35 AM, J Poz wrote: I have a specific business problem that I'm hoping someone has ideas and/or has already worked out a solution. My application needs to be able to automatically create and issue faxes to many different fax machines. The volume is going to be very high. And it is only about sending faxes and not receiving them. My application is hosted by an ASP but the Linux (Fedora 2) server is mine (dedicated). So the option of having PSTN lines to do faxes is not an option since I don't own nor can put anything in the data center. I found a SIP/VOIP provider that says they do faxing (and I can connect to them using my own device (meaning asterisk or something else if necessary)). Their requirement for faxing to work on their end is to make sure i send them via their voip service using G.711 codec. So I've done alot of research on faxing and asterisk and hylafax but I' m still at a loss. For starters, what is the architecture that I need? my application -- QUESTION MARK??? VOIP Provider --- PSTN --- Fax Machine. So first question, what should QUESTION MARK be? Is it just Asterisk or a combination of Asterisk and something like hylafax (fax manager). And depending on that answer, what is the configuration that has to be made on it. Even reference to material that explains the configuration would be very helpful to me at this time. Thanks in advance for the help, The missing link might be iaxmodem. It has two interfaces: IAX channel for asterisk, and a serial device (in /dev/) which emulates a faxmodem. Then, fax away using hylafax. I have tried faxing over SIP through a provider (broadvoice) to a coworker's fax on the pstn this way, and it worked. I haven't done any testing in volume, though. So you would have something like: Doc - hylafax - iaxmodem - * - voip provider - pstn - fax machine Phil PS- I suppose if you had multiple SIP accounts with a provider, you could create multiple iaxmodems and do things in parallel (assuming enough bandwidth and cpu). PPS- I hope you're not doing fax-spamming with this set up! ;') ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival and Asterisk - different voices?
Michael Collins wrote: Just curious to know if anyone uses Festival with * and whether or not you’ve got a different voice than the default. I’m looking at doing a commercial application but my boss doesn’t want to shell out the $ before we do some real world testing of * and Festival. Specifically, I’m looking for a female voice, preferably US English. You can change the voice by editing the asterisk function. I think you want 'voice_cmu_us_slt_arctic_hts': ;;; Command for Asterisk begin (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. ; different voices, uncomment the one you want: ;(voice_cmu_us_awb_arctic_hts) ;(voice_cmu_us_bdl_arctic_hts) ;(voice_cmu_us_jmk_arctic_hts) (voice_cmu_us_slt_arctic_hts) ;uk voices ;(voice_kal_diphone) ;(voice_ked_diphone) (utt.send.wave.client (utt.wave.resample (utt.wave.rescale (utt.synth (eval (list 'Utterance 'Text string))) 5) 8000))) ;;; Command for Asterisk end ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
Clint Sharp wrote: I'm still having numerous echo issues, even on SIP calls, with the GXP-2000s. Unfortunately, they cause echo on the remote end on SIP calls, which does not occur on other phone models. The speaker phone is unusable due to echo problems. Maybe the 1.0.2 firmware branches will help, but I'm scared of upgrading with no path back to a stable firmware. They're really nice hardware, but unfortunately the software for them just stinks (no gain control on the handsets or speakerphones, lots of missing options I'd like). Unfortunately, I have yet to find a sub $100US phone that I like. I definitely would not order 15 until I'd ordered a couple and tested. We have three of them for evaluation (fw 1.0.1.12). They are OK. Some issues we have: - Echo sort of comes and goes, but for the most part it isn't too bad. Can get rather distracting to others on conf calls, though. Seems to happen when using the handset, but not w/ speakerphone? - Speaker phone is actually quite good, but quiet - headset jack takes over the speaker phone, not the handset (which is rather odd) - handset cable is not compatible with most headset kits, it seems Price is good, and there are lots of features. Overall design is not bad. Display is OK. Web interface is good. HTTP firmware update didn't work for me, but TFTP did. I haven't yet tried the 1.0.2.x firmware. Sounds promising. Hopefully it will be 'stable' soon. Phil Clint On 2/17/06, *Mimmus* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones. - Is it a good choice (budget limit of 100 Euro/phone is mandatory)? - Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk? Thanks in advance -- Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP
Rick Smith wrote: Phil; What link ? Your question is a bit vauge, but here are some relevent urls: Sprint CoS request form (a 2 pager, with some great links to a guidelines doc and faq): http://www.sprintlink.net/maint/cos_template.cgi QoS: http://www.voip-info.org/wiki/view/QoS Phil We're got a T1 from Sprint that we use for internet. During VIOP calls, if you download something, the VOIP calls break up. I found some info at Sprint for adding 'class of service', and I also have some information on configuring our Cisco routers. I've read the relevent pages on the wiki, but it seems vauge what's required and what's required by the NSP (Sprint). How have you dealt with this problem? Is this something which requires the NSP to be involved, or can this all be done on the premises side in IOS configuration(s)? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP
David Choo wrote: Hi, Consider doing rate limiting / bandwidth reservation. It worked heaps of wonders for mine! That's good to hear. BTW- Am I doing this right? Here are the relevent chunks of my config on my router: ! ! class-map Platinum match access-group 101 ! ! policy-map IPCOS class Platinum bandwidth percent 35 ! access-list 101 permit udp any any range 16384 32768 access-list 101 permit udp any any range 6050 6060 ! interface Serial0/0 service-policy output IPCOS service-module t1 timeslots 1-24 ! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Traffic prioritization and 'class of service' for SIP
Ouch, Sprint wants $200 for the priviledge. I couldn't get approval for that yet until we are closer to switching over more lines to voip. Is it possible to do something equivelent or close without Sprint's help? It seems like they are implmenting the equivelent of: service-policy input IPCOS But, on their end (as an output). I thought about adding an 'input' on my side, but it seems like it's too late at that point since it's already traveled through the bottleneck (T1). Ideas? Phil Philip Edelbrock wrote: David Choo wrote: Hi, Consider doing rate limiting / bandwidth reservation. It worked heaps of wonders for mine! That's good to hear. BTW- Am I doing this right? Here are the relevent chunks of my config on my router: ! ! class-map Platinum match access-group 101 ! ! policy-map IPCOS class Platinum bandwidth percent 35 ! access-list 101 permit udp any any range 16384 32768 access-list 101 permit udp any any range 6050 6060 ! interface Serial0/0 service-policy output IPCOS service-module t1 timeslots 1-24 ! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Traffic prioritization and 'class of service' for SIP
We're got a T1 from Sprint that we use for internet. During VIOP calls, if you download something, the VOIP calls break up. I found some info at Sprint for adding 'class of service', and I also have some information on configuring our Cisco routers. I've read the relevent pages on the wiki, but it seems vauge what's required and what's required by the NSP (Sprint). How have you dealt with this problem? Is this something which requires the NSP to be involved, or can this all be done on the premises side in IOS configuration(s)? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Say YES to continue prompts
We're having a problem with call screening with our existing legacy system (Toshiba DK40i) which the touch-tone buttons don't work when * calls extensions. At first I had set up a 'press 1 to accept' prompt, but it won't work if the DTMF buttons aren't functioning, of course. So, a thought: Is there a way for Asterisk to listen and hear something? It doesn't have to do voice to text translations or anything so complex as that, just sense noise past a threshold and duration. Mostly, I'm just trying to sense if the user is available and at their desk. Thanks! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Auto disconnect after xx seconds of silence
I've got lines coming in from a legacy system (into FXO ports) which does not give any disconnect notification. Folks familiar with the system say that I can buy or build a device which will listen for so many seconds of dead air and then automaticly send a disconnect signal to free up any hung channels. This seems like something that could be done in software with Asterisk? Right now the main problem is that these users join a MeetMe meeting and don't free the line after they hang up. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] txfax application problem
Jeff Herring wrote:
would you care to share with the list your installation procedure
and configuration files associated with your iaxmodem and hylafax
installation alongside asterisk?
Sure! Some things, I'm sure, could use improvement, but this is working
for me:
Get iaxmodem: https://sourceforge.net/projects/iaxmodem
You need libiax2 and spandsp-0.0.3 (yes, the devel one not the other
installed. Both are included in iaxmodem (in the lib directory),
however I grabbed a slightly newer spandsp-0.0.3 from the spandsp site.
Make sure spandsp and libiax2 are found by your system (usually by doing
a 'ldconfig').
Build and install iaxmodem.
iaxmodem wants a config file at /etc/iaxmode-cfg.something. Mine looks
something like:
# cat /etc/iaxmodem-cfg.ttyIAX
device /dev/ttyIAX
port4569
refresh 60
server YOUR.SERVER.IP.HERE
peernameiaxmodem
secret YOUR_SECRET_HERE
cidname John Doe
cidnumber 8005551212
codec slinear
swapbytes true
Now, create an entry for the iax channel in your Asterisk config. Mine
looks something like this (in iax.conf):
[iaxmodem]
type=friend
username=iaxmodem
secret=YOUR_SECRET_HERE
context=faxout
host=dynamic
auth=md5,plaintext,rsa
Notice that the context is 'faxout' in my extensions.conf. Here's what
the relevent contexts are in my extensions.conf:
[fax]
exten = s,1,Dial(IAX2/iaxmodem)
[faxout]
exten = _.,1,Dial(Zap/g2/${EXTEN})
Notice I also have 'fax' which is incoming. That's a context for my zap
channel (a dedicated fax line). From zapata.conf:
group = 2
faxdetect=both
faxdetect=incoming
faxdetect=outgoing
faxdetect=yes
rxgain=0.0
txgain=0.0
context=fax
channel = 4
OK, what this all does thus far: It sets up a serial port, /dev/ttyIAX
in this case, which looks like a fax-modem that is connected to the
provided iax channel. You can point minicom at it and play with it if
you want. Calls coming into Zap-4 will automaticly go to iaxmodem and
'ring' on the /dev/ttyIAX serial device. Faxes going out on iaxmodem
automatically go our on the same Zap channel (although doesn't have to).
Now, run iaxmodem (e.g. iaxmodem ttyIAX), after you've already got
asterisk going, to get it registered. Make sure it registers and things
look OK (iax2 show peers). You could even try to call it or dial out w/
minicom (using /dev/ttyIAX as the modem device).
Now, you can set up hylafax. I installed from RPM, which was pretty
easy following the directions. Run through its set up and get the email
addresses and those relevent things set. Instead of setting up a new
modem config, however, I edited and then copied the supplied one out of
the iaxmodex distro (config.ttyIAX).
Get hylafax going (/etc/rc.d/init.d/hylafax start).
Now, here's the only stumbling block that I had: In order for things to
work, faxgetty needs to be running! The hylafax service doesn't do this
for you, you need to set it up yourself. The easiest way is to add it
to your /etc/inittab. I added it to mine like this (the new line is the
last here, the rest were already there and included for context):
# Run gettys in standard runlevels
1:2345:respawn:/sbin/mingetty tty1
2:2345:respawn:/sbin/mingetty tty2
3:2345:respawn:/sbin/mingetty tty3
4:2345:respawn:/sbin/mingetty tty4
5:2345:respawn:/sbin/mingetty tty5
6:2345:respawn:/sbin/mingetty tty6
7:2345:respawn:/usr/sbin/faxgetty ttyIAX
You may need to restart or 'telinit q' or something to get the changes
noticed by init.
Now you can try sending and receiving faxes. For fun, I created a new
account on the Asterisk server and put a .procmailrc file there which
passes emails to that account to hylafax:
SUBJECT=`formail -xSubject:`
:0 c
* ^Subject: [EMAIL PROTECTED]
|/usr/bin/faxmail -d $SUBJECT
How it works: send a mail to the account and in the subject line put
[EMAIL PROTECTED], and it will fax the email to 555-1234 with 'attn Joe' on
the cover page. Slick... although it seems to take more work to convert
non-text (e.g. images), which I haven't attempted yet.
OK, so there's the crash course. I hope it helps.
Phil
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Re: [Asterisk-Users] txfax application problem
iaxmodem + hylafax worked much better for me. Seems solid where txfax/rxfax was very iffy. Thus far, I'm just using it with Zap lines, though. Phil Technical Support wrote: Downgrade your spandsp. Do some reading on spandsp first! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Allan Gee Sent: Tuesday, January 24, 2006 10:36 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] txfax application problem Nobody seems to use txfax or does nobody have any problems with it? I have sent mails to most lists and get no reply. I cannot get a fax to go through with txfax. I use a call file as a test and all I get on the receiving fax is a bunch of vertical lines. my call file is: Channel:Srx/gout/4658158 MaxRetries: 0 WaitTime: 20 Application:txfax Data:/etc/asterisk/testfax.tif|caller|debug FYI tiffinfo on file is: TIFF Directory at offset 0x733e Image Width: 1728 Image Length: 1161 Resolution: 77, 38.5 pixels/cm Bits/Sample: 1 Compression Scheme: CCITT Group 3 Photometric Interpretation: min-is-white FillOrder: lsb-to-msb Date Time: 2006/01/11 18:06:14 Host Computer: asterisk.equation.co.za Orientation: row 0 top, col 0 lhs Samples/Pixel: 1 Rows/Strip: (infinite) Planar Configuration: single image plane Page Number: 0-1 Software: spandsp Group 3 Options: 2-d encoding+EOL padding (5 = 0x5) Fax Data: clean (0 = 0x0) Fax Receive Time: 29 secs what is going on here? I have used latest spandsp-0.0.3pre23 with it's app_txfax.c and tiff-3.7.1 from source I have also used just about every other spandsp and libtiff combo but no joy with sending the faxes. PLEASE HELP!! Regards Allan Gee Phone: +27 21 4644400 Ext. 103 www.equation.co.za ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
Kristof Hardy wrote: Was there a resolution to this issue? The GXP-2000 seems to be a very popular phone, so I can't imagine others on the list not experiencing this? Or is this part of a batch with unresolvable problems that I need to send back to the seller? Well, I'm using dozens of these phones without this problem. What kind of DHCP/ntp server are you using? I'm using dnsmasq on a Debian box, together with the ntp-server. I'm using a mixture of 1.0.1.13 beta and .12 firmwares, both working correct. The DHCP server is on the same 100BaseT switch as the phone right now (they are literally just a few feet away from each other). DHCP server is on Fedora 3 Linux Internet Systems Consortium DHCP Server V3.0.1 (from the rpm: dhcp-3.0.1-44_FC3). Packet sniffer shows the phone getting in some sort of fight with the dhcp server. I attached a text dump of the sniff. You can see a repeating conversation from packet 20 to 40, and it continues on and on like that. And, my logs are filling up with gazillions of these (pattern repeats every 3 seconds): Jan 23 12:06:41 DrTheopolis dhcpd: DHCPDISCOVER from 00:0b:82:05:a9:bf via eth0 Jan 23 12:06:41 DrTheopolis dhcpd: DHCPOFFER on 206.228.191.144 to 00:0b:82:05:a9:bf via eth0 Jan 23 12:06:41 DrTheopolis dhcpd: DHCPREQUEST for 206.228.191.144 (206.228.191.7) from 00:0b:82:05:a9:bf via eth0 Jan 23 12:06:41 DrTheopolis dhcpd: DHCPACK on 206.228.191.144 to 00:0b:82:05:a9:bf via eth0 While I was thinking of logs, I set up remote syslog for the phone, but all I see while it is set to dhcp is a single log noting the firmware versions on the phone. With a static IP it logs info about registering w/ * (which it does successfully and I can make calls). Phil 1 0.00 0.0.0.0 - 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xaabbccdd 2 0.727622 206.228.191.7 - 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xaabbccdd 3 0.746653 0.0.0.0 - 255.255.255.255 DHCP DHCP Request - Transaction ID 0xaabbccde 4 0.749231 206.228.191.7 - 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xaabbccde 5 0.766593 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.1? Tell 206.228.191.144 6 0.997865 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.1 is at 00:10:4b:96:2f:eb 7 1.308918 206.228.191.144 - 206.228.191.7 DHCP DHCP Release - Transaction ID 0xaabbccdf 8 14.164223 0.0.0.0 - 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xcecb 9 14.164531 206.228.191.7 - 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xcecb 10 14.166809 0.0.0.0 - 255.255.255.255 DHCP DHCP Request - Transaction ID 0xcecc 11 14.172534 206.228.191.7 - 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xcecc 12 14.175408 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144? Gratuitous ARP 13 14.339375 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 14 17.155641 206.228.191.144 - 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xcece 15 17.155975 206.228.191.7 - 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xcece 16 17.158134 206.228.191.144 - 255.255.255.255 DHCP DHCP Request - Transaction ID 0xcecf 17 17.159263 206.228.191.7 - 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xcecf 18 17.161118 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144? Gratuitous ARP 19 17.609869 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 20 20.155260 206.228.191.144 - 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced0 21 20.155760 206.228.191.144 - 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xced1 22 20.155981 206.228.191.7 - 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xced1 23 20.158255 206.228.191.144 - 255.255.255.255 DHCP DHCP Request - Transaction ID 0xced2 24 20.159714 206.228.191.7 - 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xced2 25 20.161242 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144? Gratuitous ARP 26 20.640088 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 27 23.165159 206.228.191.144 - 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced3 28 23.165658 206.228.191.144 - 255.255.255.255 DHCP DHCP Discover - Transaction ID 0xced4 29 23.165879 206.228.191.7 - 206.228.191.144 DHCP DHCP Offer- Transaction ID 0xced4 30 23.168148 206.228.191.144 - 255.255.255.255 DHCP DHCP Request - Transaction ID 0xced5 31 23.170237 206.228.191.7 - 206.228.191.144 DHCP DHCP ACK - Transaction ID 0xced5 32 23.172210 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144? Gratuitous ARP 33 23.180374 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 34 26.165097 206.228.191.144 - 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced6 35
Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP (SOLVED)
Tony Hoyle wrote: Philip Edelbrock wrote: 18 17.161118 Grandstr_05:a9:bf - BroadcastARP Who has 206.228.191.144? Gratuitous ARP 19 17.609869 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144 is at 00:10:4b:96:2f:eb 20 20.155260 206.228.191.144 - 206.228.191.7 DHCP DHCP Decline - Transaction ID 0xced0 It looks like your DHCP server is in fact broken. It's passing out duplicate addresses - the device 00:10:4b:96:2f:eb already has 206.228.191.144, so the Grandstream (correctly) declines the offer. The server then tries to send the same address *again* instead of selecting a new one, and the same sequence ensues. It should give a different address if the original one is declined. Ah, you are close! I figured it out (*hurray!*). It was in fact a misconfiguration on my part. 144 isn't the end of my subnet, 143 is. So, packet 18 is the phone confirming that it owns IP 144. Packet 19 is from the router saying, no you don't, I own that (this is a proxy arp setup). So, the phone declines and requests a new IP. The head scratcher was that for the next request, it requests 144 again, so the DHCP server says (again) OK, you got it and the loop continues. Once I adjusted my dhcp config to end my dynamic pool at 143 instead of 144, all was well. Additionally, I noticed that the phone requests these pieces of info in the dhcp response: - Subnet - Router - DNS server(s) - Time Server(s) --- !! So, I additionally put in the dhcp config a time server (the ip for time.nist.gov for now). And after the first reboot, the phone gets an IP, pings the dhcp server once, registers, sets it's time, checks for firmware updates, and seems perfectly happy. Hurray! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GXP-2000 fw 1.0.1.13 and NTP
On Dec 31, 2005, at 7:28 AM, Ross C wrote: Peter, After upgrading to 1.0.1.13 I had some miscellaneous problems on one of my GXP-2000's--it would grab an IP address, but it wouldn't get the time/date, it wouldn't register, blah blah blah. I could access the web interface OK, so it wasn't a network issue (I don't think). Anyway...I ended up resetting to factory defaults and all is well now. Maybe try that? That has solved some other problems I've had as well. I just got a 2000 which does exactly this (our first for evaluation.. which is somewhat disappointing thus far). I could see in a packet sniffer a weird cycle of DHCP requests like it got an IP but kept retrying? A power cycle doesn't solve the problem (it's had many, and dozens of software resets). A reset with the MAC input doesn't work either for me. The phone was at an older FW when I got it (ending in .9, I think) and then updated to to the latest stable (.12 I think off the top of my head). Btw- the firmware update was a pain. HTTP updates were hitting the server (Apache) with 'bad request' results. I needed to set up my own tfpt server to make it work. Off lan updates weren't working, either, in any case. The phone will register and work when it has a static address assigned, but not when set for DHCP. In all cases, the clock is always wrong. I can see with a packet sniffer that the NTP request is sent and received, but with no effect on the phone display. Was there a resolution to this issue? The GXP-2000 seems to be a very popular phone, so I can't imagine others on the list not experiencing this? Or is this part of a batch with unresolvable problems that I need to send back to the seller? Thanks! TGIF! :') Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P zttest not working
Random thought: They look like they are owned by asterisk. Are you running zttest under an asterisk account or as root? Phil Antonio Moragues wrote: The device is in place: # ls -l /dev/zap/ total 0 crw-rw 1 asterisk asterisk 196, 1 Jan 19 23:41 1 crw-rw 1 asterisk asterisk 196, 2 Jan 19 23:41 2 crw-rw 1 asterisk asterisk 196, 3 Jan 19 23:41 3 crw-rw 1 asterisk asterisk 196, 4 Jan 19 23:41 4 crw-rw 1 asterisk asterisk 196, 254 Jan 19 23:40 channel crw-rw 1 asterisk asterisk 196, 0 Jan 19 23:40 ctl crw-rw 1 asterisk asterisk 196, 255 Jan 19 23:40 pseudo crw-rw 1 asterisk asterisk 196, 253 Jan 19 23:40 timer On 1/20/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jan 19, 2006 at 11:53:37PM +0100, Antonio Moragues wrote: Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS1 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ¿anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Some info: # lsmod Module Size Used by wctdm 35264 1 zaptel188804 7 wctdm binfmt_misc12040 1 The module was loaded. Is the device file in place? ls -l /dev/zap/pseudo -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Ben Fried wrote: Just to be clear, I have inbound and outbound faxes working with my TDM400, by going the iaxmodem and hylafax route. No need for a separate modem or an x100p card. Be Ah, that's interesting. Can you provide some details on how you set it up? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Ben Fried wrote: On 1/9/06, Rich Adamson [EMAIL PROTECTED] wrote: Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is terminated without successful transmission of the fax. I get logs that look what's included below. From reading the list, it looks like this is caused by the TDM card missing frames. Does that sound correct? If so, is there any relief in sight? Its been a problem since the card came out a couple of years ago. So, no it does not appear there is any relief in sight. Sigh. What a disappointment! Are there any other options for home users to receive faxes over the PSTN through *? Is anyone working on an alternative to the zaptel driver that might fix this issue? Humm, I tried to get my TDM400 card accepting faxes last week. It works about 1 out of 8 times. When it works, it looks great. When it doesn't, I usually (but not always) get a 'poor line quality' error from the sending fax machine and a blank or small corrupt image. I've tried adjusting the gain up and down, reversing ring/tip, and a few other little things. I wonder if it helps to adjust some other settings in the zapata.conf, like echo cancellation? My hope is dwindling, though, after reading this thread. :'( I do have a couple unused X100P clones sitting in that box that might be worth a try... Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp, rxfax, TDM400/zaptel, missed frames, any help?
Ben Fried wrote: Since writing my message, I appear to have had success using iaxmodem + hylafax to do inbound faxing. Setup was not completely obvious, especially if you're usin [EMAIL PROTECTED], like me, I finally seem to have inbound faxes working properly now - 5 or 6 in a row have all come in just fine. Ben Incidentially, I switched to a X100P and it works a lot better. That is to say that one fax failed of 5. Not great, but much better. We've got a semi-nice (but old) fax/modem from the days when we dialed in from home. If we get serious, I might try that instead. Or just stick to smearing ink/toner on dead trees. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS or VOIP
Jim Freeze wrote: [...] So for 5 phones, I would need 2 cards. And, the O'Reilly book says that I should not put 2 cards in the same box, so I would need another computer. [...] Whoa, I'm confused. Can't you use as many cards as you have slots? We've got just one 4-port card, but I've always assumed it was just a matter of purchasing and installing more to get 8 or 12 lines? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup Detection (revisited)
Darrick Hartman wrote: A little background. I'm integrating asterisk as the voicemail service for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2 is used to connect an analog device (such as a voice modem) to the pbx. In the past we've used vgetty and a voice modem with varying degrees of success. If you haven't yet, I'd turn on busydetect in zapata.conf. Can't hurt and might (although unlikely) work (I had to turn it on to make it work on my system). Switching to loop-start might be worth a try, too. For a while my VM * system wasn't doing disconnect detection, and it was OK. I had trouble with the single-port cheapo cards off eBay with the silence thresholds, but using a TDM400P card fixed that for me. Also make sure all you menus will time out and hang up. You could try posting to the Nortel list: http://www.tgrace.com/mailman/listinfo/nortel-list They've been very helpful and kind to me. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
Lots of great comments. Got me doing a lot of researching and thinking. What I love is the ability to mix and match IAX and SIP devices on the same system. A breath of fresh air compared to our existing Toshiba key system. That said, I think we'll be doing a combination of things. Regular SIP hardphones, atas, and sip wifi phones. I'm still a bit stuck on the wifi phones because we can have folks use them in the office, the coffee shop, and at home (where ever wifi is, basicly). Cost is pretty similar, really, compared to an ata and cordless setup (approx $200 either way). I might wait for the F3000. It seems to support better security, networking (g), and has a nicer profile. Phil Paul Mahler wrote: I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless phone. It will be less expensive and it will likely work better. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joash Herbrink Sent: Monday, January 09, 2006 11:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Recommendations on a WiFi phone for *? The zyxel p2000W Works fine, good batt. Live. Decent sound quality. All in all a good product for about 150 euro's -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philip Edelbrock Sent: Tuesday, January 10, 2006 2:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Recommendations on a WiFi phone for *? We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.15/223 - Release Date: 1/6/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SOLVED: Hung Zap channels connected to old key system
We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it. Some proprietary voicemail systems (and probably tie-lines, too) like to use DTMF tones instead of standard ground/loop/kewl whatever signaling. Our key system was programmed to use such DTMF tones instead of the usual analog signaling on those ports. (I think it was program 31 on our Toshiba DK40i) Asterisk of course ignored those, but the other systems used those for line signaling (including our previous 3rd party system). Amusingly, I know now why for years we kept hearing loud DTMF tones when our branch office picked up their phones. Their system, too, was configured to have those analog lines be connected to a voicemail system and not to a FXO port on a T1 CSU. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommendations on a WiFi phone for *?
We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording Calls at the phone
Michael Sampson wrote: I'm not really trying to monitor anything on the asterisk box at all. I guess this is more of an SIP phone question. Really all I need is to get the audio from an SIP phone, both the caller and callie, to a 1/8th inch stereo jack that I can plug into a mic input. Another possible option, if it helps, is running Cain and Abel on a PC connected to the same network as the SIP phone. It automaticly spools voip calls on the network to wav files. http://www.oxid.it/cain.html Phil Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Douglas Garstang wrote: On Demand-monitoring? If your referring to monitoring specific agents calls, I'm still trying to work out how to do that. You can either monitor all calls for a queue, or all calls for all agents, but not all calls for a specific agent. I tried to use the Monitor() command on it's own to start recording when an agent receives a call, but that does not appear to work. -Original Message- From: Francesco Peeters (Asterisk) [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 7:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Recording Calls at the phone On Fri, January 6, 2006 15:37, Michael Sampson said: I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do that. I could get an FXO adapter and regular phones, but I'm looking to get as little equipment as possible. Radio shack makes a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. Asterisk has a built in monitoring system. You can chose to do Always, Never or On Demand monitoring, depending on your setup and dialplan http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good luck! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Integrating with Toshiba Strata DK40i KSU
We've done a direct swap of an old Amanda voicemail system with a shiney new Asterisk system (Asterisk 1.0.9). The system consists of 4 FXO ports on the * box (TDM400P), and three old Wildcards we aren't using (too buggy we found). CO lines- Toshiba - FXO ports on * We want to branch out a little more and use it as an auto-attendant. The first problem seems to be an asterisk problem. When ringing extensions, it thinks the first ringback is an answer: == CDR updated on Zap/7-1 -- Executing Macro(Zap/7-1, dialexten|35) in new stack -- Executing Dial(Zap/7-1, Zap/6/351|5|m) in new stack -- Called 6/351 -- Started music on hold, class 'default', on Zap/7-1 -- Zap/6-1 answered Zap/7-1 -- Stopped music on hold on Zap/7-1 -- Attempting native bridge of Zap/7-1 and Zap/6-1 To the caller, they hear on-hold music for just a brief second, and then ringing. When they hang up, the lines remained bridged and the extension continues to ring until I log in and do some 'soft hangup' commands. The second problem is more of a Toshiba problem (or rather my lack of knowledge of). I hope that perhaps somebody might be able to help me? I want to have a way to ring multiple extensions if sombody, say, hits zero. The Toshiba can ring mutliple extensions for fresh new incoming calls, but once answered I can't seem to 'unanswer' the call to get it ringing at multiple stations (we have no designated reception phone that is staffed 100% of the time). Thanks! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream web configuration utility
Ron Bulthuis wrote: I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386. Browsing to each device by IP address, I can get logged in using admin and I can see the advanced settings, however, if I try to change the settings and clicking the Change button, it just brings me back to ask for the password again.. Two quick thoughts: Try a different browser, and make sure your security settings on your browser/PeeCee aren't too strict. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Passing authentication to an analog adapter
This is more of a curiosity and a thought than serious issue. But, I wonder if I can get my Asterisk server to authenticate to my provider by throwing the authentication requests to the SIP analog-adapter they shipped me? (And I can't get in and see the authentication credentials in the adapter, of course.) In other words, say I've got something like: SIP analog adapter -- * server -- provider Such that the DHCP and DNS information the SIP adapter gathers comes from the asterisk server where it pretends to be the provider. So then this happens: - * tries to register with the provider - gets the challenge - the challenge is passed to the SIP adapter - which answers the challenge correctly - then * passes the answer to the provider and completes the registration (AKA man-in-the-middle) Is there such an authentication feature in *? If not, I could see it being handy. An alternative (although not as fun) would be to connect the analog adapter to a FXO card in the * server. PS- I know some providers could treat this as a violation to their service agreement, btw. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Affordable IP Phones for Asterisk
We're using a Budgetone 101 ($60) SIP phone. It works pretty well. No echo cancellation, though, which is a little annoying when used somewhere with significant ping-times to the server. Phil Rehan Ahmed wrote: Hello Dakota, I have a few that i can ship you from vida21.com http://vida21.com for 70$ each they with me in the US. The client is now using cisco i have 70 pcs with me Rehan On 12/20/05, *Dakota* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Are there any IP Phones that can work with Asterisk, that cost less than $60 ? if so, what's the model/manufacturer? Dakota ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today. http://www.didx.net - DID Number Exchange and Peering Service. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and echo cancel
On Dec 18, 2005, at 12:01 PM, Andrew Kohlsmith wrote: On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote: As a matter of fact im serious to know where is the source of echo in a pure VoIP connection, i think the most of echo problems come from hybrid circuits which are not an issue in pure VoIP sessions. Easy. Get better endpoints. In a pure-voip loop you have echo due to acoustic coupling from the earpiece to the mic, or the speaker to the mic in a speakerphone. Easy way to tell: in a call with bad echo, have the other side mute. If your echo goes away, you've got your culprit. Also note that if your transmit level is too high or they have the volume up too loud on their end it could push the audio coupling over what the design specifications were. We're having some issues with a Budgetone, especially in speaker phone mode causing echo. I think I read the specs have a feature line item of Echo cancellation (pending), lol. No way to fix this other than buying new phone(s)? Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] looking for hardphone configuration info
Bruce Ferrell wrote: Hi all, I just aquired some new SIP phones as gifts from a friend, a Uniden UIP200 and UTstarCom Wifi F1000. Unfortunately neither came with information about how to configure them remotely. From what I see on the UTstarCom user forums if the phone comes from voipsupply (it did) they are supposed to provide the requsite info but seem to drag their feet on doing that (I've not yet contacted them) I found config info on the uniden phone on voip-info.org, but it would be nice to see a reference on what's there Help anyone! An FCC ID search can get you info on the phones (including on the new F3000 which just showed up about 2 weeks ago): https://gullfoss2.fcc.gov/prod/oet/cf/eas/reports/GenericSearch.cfm Punch in the FCC ID (which must be somewhere on the phone) and you'll get a list of downloads, usually including an PDF manual. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - No DTMF tones
Sorry, this is slightly off topic, but I wonder if somebody has some hints on getting our Meridian system to output DTMF tones to our Asterisk box. Simply put, when buttons as pressed, nothing happens. The Asterisk box has a 4 port Digium FXO card. This is what we've got: Meridian M8X24-DS Meridian M12X0 Thanks for any tips! Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting Meridian M8x24-DS to Asterisk - No DTMFtones
On Dec 15, 2005, at 6:23 PM, Steve Totaro wrote: Sorry, this is slightly off topic, but I wonder if somebody has some hints on getting our Meridian system to output DTMF tones to our Asterisk box. Simply put, when buttons as pressed, nothing happens. The Asterisk box has a 4 port Digium FXO card. This is what we've got: Meridian M8X24-DS Meridian M12X0 Thanks for any tips! Phil Hey, thanks for the response. I have another system (Toshiba) which works fine with it, so I'm doubting that it's a issue specifically with the Asterisk server or it being able to properly decode the tones. Have you put a butt set on the line and listen to see if there is DTMF and it is just not being recognized? I've done similar tests where instead of the asterisk server, I used a phone instead. When buttons are hit, no tones are emitted from the Meridian, but a little acknoledge beep on the Meridian phone is heard as buttons are pressed. It seems to me like a programming or some other issue on the Meridian. But, I don't know where to start to correct it. The tech is completely MIA after my call to set up an appointment. I might have to call another service group. I'm fairly competent at managing and building out Toshiba DK systems (I maintain two DK40's), but this Meridian was sort of inherited from another source. Phil ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + WiFi Phones
I'm curious if anything new has been determined on this phone? Is it SIP compatible with Asterisk and, say, Broadvoice? I'm a little wary that this may be vaporware. The phone doesn't seem to be listed by the FCC. But, I would preorder one if it's Asterisk and Broadvoice compatibile. Phil PS- Contact us form on the viopsupply site seems to be broken? Just spins for me. Cory Andrews wrote: The F3000 is also a clamshell, flip type phone. I should be receiving an eval unit shortly and will post my findings after we work it over in the lab. Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Luki wrote: UTStarCom has the F3000 coming in December, which will have according to their spec * WEP (64 and 128 bit )/WPA/MD5 Auth * Handover/Roaming between different AP and SSID So what else is different compared to the F1000? The 1000 also does WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth, but SIP nonce/MD5 response certainly is implemented. Roaming kind of works, but could be improved. In one place I made it from 4th floor - elevator - lobby while on the phone and without any noticeable dropouts (ulaw codec). But the building was covered with access points, on average NetStumbler saw 6 at the same time. So it works, but not always. Don't get me wrong, the phone does have issues and in my opinion is not production quality, meaning it will freak out unexpectedly and only a reboot helps, which hardly ever happens to any Sipura adapters or phones. Hopefully the new 3.6 firmware performs better. Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hobby box
Logan wrote: Hi everyone! Okay. I was reading on the voip-info.org about FXO and FXS. Is it possible just to get a card with FXO and FXS together? I know Digium sells them, but as I've said, I'm looking to spend too much. Thanks for everyone's input! Logan. FXO is easy, but FXS is more expensive. You'll likely need two cards (one for each). You can get $10 FXO cards on ebay, but something seems to be buggy as heck with those. I have problems with them nearly daily which requires reboots. You could try finding an Internet Phonejack (I think that's the FXS one). I bought one a while ago and it wasn't too expensive (compared to the Digium stuff). Not sure if the company exists any more. Phil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P troubles?
Tzafrir Cohen wrote: [...]wcfxo is the driver for the X100P cards. FYI-I just had another crash. This time I got an oops dump: [ cut here ] kernel BUG at mm/rmap.c:493! invalid operand: [#1] Modules linked in: loop wcfxo(U) zaptel(U) crc_ccitt ipv6 parport_pc lp parport autofs4 it87 eeprom i2c_sensor i2c_isa sunrpc video button battery ac ohci_hcd i2c_sis630 i2c_core snd_trident gameport snd_ac97_codec snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd_page_alloc snd_util_mem snd_mpu401_uart snd_rawmidi snd_seq_device snd soundcore 8139too mii dm_snapshot dm_zero dm_mirror ext3 jbd dm_mod CPU:0 EIP:0060:[c018dc74]Not tainted VLI EFLAGS: 00010286 (2.6.13-1.1532_FC4) EIP is at page_remove_rmap+0x36/0x40 eax: ebx: c6800924 ecx: c04eaa78 edx: c13aa240 esi: c13aa240 edi: 0020 ebp: 00249000 esp: ce0d4e60 ds: 007b es: 007b ss: 0068 Process gawk (pid: 8941, threadinfo=ce0d4000 task=d994e000) Stack: c01814c4 0025e000 c04eaa78 d635d000 0025e000 0025e000 0025dfff c018165a 0025e000 c04eaa78 0001a000 0025e000 d8c51754 0040 c018182f 0025e000 d13d8f8c c10d0260 06813067 b7fe3000 c01814d6 Call Trace: [c01814c4] zap_pte_range+0xd6/0x1e9 [c018165a] unmap_page_range+0x83/0xb7 [c018182f] unmap_vmas+0x1a1/0x45d [c01814d6] zap_pte_range+0xe8/0x1e9 [c018a32b] exit_mmap+0x12e/0x36d [c0179b2e] __pagevec_lru_add_active+0x31e/0x47c [c01241c5] mmput+0x25/0x317 [c012d5b4] do_exit+0xe0/0x942 [c0164c71] audit_syscall_entry+0x130/0x15e [c012df6d] do_group_exit+0x12b/0x349 [c0109b52] do_syscall_trace+0xef/0x123 [c0104465] syscall_call+0x7/0xb Code: 08 ff 0f 98 c0 84 c0 75 01 c3 8b 42 08 83 c0 01 78 19 ba ff ff ff ff b8 10 00 00 00 e9 5e 30 fe ff 0f 0b ea 01 9e 74 41 c0 eb d3 0f 0b ed 01 9e 74 41 c0 eb dd 55 57 56 53 83 ec 24 89 c7 89 d3 3Debug: sleeping function called from invalid context at include/linux/rwsem.h:43 in_atomic():1, irqs_disabled():0 [c012a4e3] profile_task_exit+0x13/0x48 [c012d4ef] do_exit+0x1b/0x942 [c0104622] common_interrupt+0x1a/0x20 [c01051fa] die+0x2e5/0x3bd [c011edd3] fixup_exception+0xb/0x28 [c0105516] do_invalid_op+0x0/0xab [c01055b8] do_invalid_op+0xa2/0xab [c018dc74] page_remove_rmap+0x36/0x40 [c016c4cf] generic_file_buffered_write+0x37d/0x616 [c0170778] __alloc_pages+0xe7/0x3ff [c0131f9d] current_fs_time+0x4e/0x69 [c010467f] error_code+0x4f/0x54 [c018dc74] page_remove_rmap+0x36/0x40 [c01814c4] zap_pte_range+0xd6/0x1e9 [c018165a] unmap_page_range+0x83/0xb7 [c018182f] unmap_vmas+0x1a1/0x45d [c01814d6] zap_pte_range+0xe8/0x1e9 [c01055b8] do_invalid_op+0xa2/0xab [c018dc74] page_remove_rmap+0x36/0x40 [c016c4cf] generic_file_buffered_write+0x37d/0x616 [c0170778] __alloc_pages+0xe7/0x3ff [c0131f9d] current_fs_time+0x4e/0x69 [c010467f] error_code+0x4f/0x54 [c018dc74] page_remove_rmap+0x36/0x40 [c01814c4] zap_pte_range+0xd6/0x1e9 [c018165a] unmap_page_range+0x83/0xb7 [c018182f] unmap_vmas+0x1a1/0x45d [c01814d6] zap_pte_range+0xe8/0x1e9 [c018a32b] exit_mmap+0x12e/0x36d [c0179b2e] __pagevec_lru_add_active+0x31e/0x47c [c01241c5] mmput+0x25/0x317 [c012d5b4] do_exit+0xe0/0x942 [c0164c71] audit_syscall_entry+0x130/0x15e [c012df6d] do_group_exit+0x12b/0x349 [c0109b52] do_syscall_trace+0xef/0x123 [c0104465] syscall_call+0x7/0xb Fixing recursive fault but reboot is needed! scheduling while atomic: gawk/0x0001/8941 [c03ff83e] schedule+0x6ee/0x938 [c0104465] syscall_call+0x7/0xb [c014b526] __kernel_text_address+0x1c/0x27 [c0104b79] show_trace+0x2a/0x78 [c0104465] syscall_call+0x7/0xb [c012dbcb] do_exit+0x6f7/0x942 [c01051fa] die+0x2e5/0x3bd [c011edd3] fixup_exception+0xb/0x28 [c0105516] do_invalid_op+0x0/0xab [c01055b8] do_invalid_op+0xa2/0xab [c018dc74] page_remove_rmap+0x36/0x40 [c016c4cf] generic_file_buffered_write+0x37d/0x616 [c0170778] __alloc_pages+0xe7/0x3ff [c0131f9d] current_fs_time+0x4e/0x69 [c010467f] error_code+0x4f/0x54 [c018dc74] page_remove_rmap+0x36/0x40 [c01814c4] zap_pte_range+0xd6/0x1e9 [c018165a] unmap_page_range+0x83/0xb7 [c018182f] unmap_vmas+0x1a1/0x45d [c01814d6] zap_pte_range+0xe8/0x1e9 [c018a32b] exit_mmap+0x12e/0x36d [c0179b2e] __pagevec_lru_add_active+0x31e/0x47c [c01241c5] mmput+0x25/0x317 [c012d5b4] do_exit+0xe0/0x942 [c0164c71] audit_syscall_entry+0x130/0x15e [c012df6d] do_group_exit+0x12b/0x349 [c0109b52] do_syscall_trace+0xef/0x123 [c0104465] syscall_call+0x7/0xb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hobby box
Logan wrote: I was wondering if it was feasable to istall Asterisk on this box and have that modem (or whatever modem) with a regular telephone wired to the Phone port. I'm a bit of a noob, also, but I don't think the Phone port on those cards are real FXS ports. I.e., I think they just connect through to the PXO jack while the modem is not in an off-hook state. Can someone verify this? A test might be to see if you get 'battery' (voltage) on the phone port when nothing is connected to the line/fxo port? Phil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P troubles?
I've got a voicemail server I made from four X100P cards (off eBay), Fedora Core 4, connected to a Toshiba DK40 system. I'm using Asterisk 1.0.9, and Zaptel 1.0.9.2. It works great, except the card which receives a majority of the activity occationally will go into a 'Red' alarm and then Asterisk won't answer calls on any of the cards, eventhough the other cards are not in an alarm state. If I reboot the server, all it well again. I've switched around the ports in case of a hardware issue, but the same thing happened on a different card (again, it was handling a majority of the calls). I don't see anything in the logs, but I may not know what to look for. Is this a known issue? Thanks! Phil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P troubles?
On Nov 13, 2005, at 4:31 PM, Noah Swint wrote: Are you running off the rpms or compiled version? Compiled. Actually, I had to compile and install it twice because the first time I didn't have Zaptel installed (which needs to be installed first, apparently). Do you suppose it makes a difference that I'm not using the RPMs? Phil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P troubles?
On Nov 13, 2005, at 6:09 PM, Rich Adamson wrote: About a year and a half ago when I was running a couple of x100p's there was an issue associated with disconnecting the pstn line from the card. If I recall correctly, if the pstn line was removed for more then a second or so (a couple of times), the card would go into some unknown state and it had to be restarted. The comments/fix at the time were oriented around don't do that instead of fixing the code to recognizing it. I don't have a clue whether that is still an issue with the x100p code or not. Humm, when I moved the jacks around (trying to get the 'bad' card out of the loop) the machine hung. You might try using a regular old voltmeter on the tip/ring of the line going to the Toshiba and watch what happens at the end of a VM call. It is possible the Toshiba is opening/disconnecting the line for some rather lengthy period (disconnect supervision). I fear you might be right. I used zttool a few times while the machine was in production. At least once I saw the card which gets the most use in a 'red' state and then switch back to OK when activity (ringing) occured. Sigh, looks like a hardware/driver issue that's been around for at least a year and a half?! Say it isn't so. :'( Phil ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
